한국어

소프트스위치

온누리070 플레이스토어 다운로드
    acrobits softphone
     온누리 070 카카오 프러스 친구추가온누리 070 카카오 프러스 친구추가친추
     카카오톡 채팅 상담 카카오톡 채팅 상담카톡
    
     라인상담
     라인으로 공유

     페북공유

   ◎위챗 : speedseoul


  
     PAYPAL
     
     PRICE
     

pixel.gif

    before pay call 0088 from app


https://jitsi.org/GSOC2010/Kamailio4575Accepted


http://opensips-open-sip-server.1449251.n2.nabble.com/No-Voice-Comm-in-Conference-call-td7580232.html


http://www.in2eps.com/fo-sip/tk-fo-sip-service-11.html


http://wiki.cs.columbia.edu/download/attachments/576/SIP+Conferencing.pdf

GSoC Student: Marius-Ovidiu Bucur - (Romania) 
Mentors: Daniel-Constantin Mierla (Romania/Germany) 

PROJECT REQUIREMENTS ( SHOW )

In case you’ve already participated in conference phone calls (which are basically confs with many participants) then you most probably had to simply dial a number and then somehow started hearing everyone. This is how things have been happening in conventional telephony for quite a while and this is how they happen today with VoIP.

In the case of VoIP, however, the approach is not all that sophisticated since VoIP clients would have the impression they are calling a regular participant and they would hence present you with their regular call interface. This works of course, but why settle for it when we could have more :). Wouldn’t it be nice for example if you could see who else is on the call? Wouldn’t it be even better to know who’s currently speaking?

We think this is important and so do the members of the popular Kamailio (OpenSER) development team. We are therefore joining up in this project and need your help to add the necessary code to Kamailio.

kamailio.png

In the SIP specification universe (or in other words in the IETF), conference calls are described by RFC 4353, and RFC 4575. The basic differences between these two are explained in these slides but you’d still need to have a look at the specs :).

So to sum it up, this project is about the implementation of conference signalling in the Kamailio (OpenSER) server. It means implementing support for the following standards:

  • RFC 4353: A Framework for Conferencing with SIP
  • RFC 4575: A SIP Event Package for Conference State

Interested? Then looking forward to reading your application!

Note that this project will be mentored by members of the Kamailio (OpenSER) development team so you’ll have all the expert help you need!

References:

Kamailio (OpenSER) – the Open Source SIP Server
http://kamailio.org

A SIP Event Package for Conference State
http://tools.ietf.org/html/rfc4575

A Framework for Conferencing with SIP 
http://tools.ietf.org/html/rfc4353

Support for conference calls in SIP Communicator
http://sip-communicator.org/gsoc2010/SIP.Communicator@FOSDEM-2010-02-06-updated.pdf

Other Jitsi GSoC Projects 
http://gsoc.jitsi.org

Jitsi Developer Documentation
http://www.jitsi.org/index.php/Documentation/DeveloperDocumentation

The official Jitsi website 
http://www.jitsi.org

조회 수 :
94275
등록일 :
2014.03.12
12:31:17 (*.251.139.148)
엮인글 :
http://webs.co.kr/index.php?document_srl=39231&act=trackback&key=063
게시글 주소 :
http://webs.co.kr/index.php?document_srl=39231
List of Articles
번호 제목 글쓴이 날짜 조회 수
52 Video conference server OpenMCU-ru - Introduction admin 2014-04-01 59987
51 SIPSorcery admin 2014-03-18 49012
50 Ekiga (formely known as GnomeMeeting) is an open source SoftPhone admin 2014-03-12 46906
49 telepresence: Open Source SIP Telepresence/MCU admin 2014-03-12 196420
48 SIP PBX - OpenSIPS and Asterisk configuration admin 2014-03-12 175512
» Conference Support in Kamailio (OpenSER) admin 2014-03-12 94275
46 OpenSIPS configuration for 2 or more FreeSWITCH installs admin 2014-03-12 80203
45 The Impact of TLS on SIP Server Performance file admin 2014-03-12 47126
44 book-opensips-101 / content / 3.2. SIP TLS Secure Calling.mediawiki admin 2014-03-12 47219
43 Where to check OpenSIPS does not start? admin 2014-03-09 48126
42 opensips-1.10.0_src.tar.gz experimental source code documentation admin 2014-03-09 42290
41 Kamailo OpenSIPs installation on Debian admin 2014-03-09 90382
40 Using the openSIPS Registrant Module admin 2014-03-09 56668
39 RTPproxy Frequentry Asked Questions (FAQ) ¶ admin 2014-03-07 183495
38 Building Telephony Systems with OpenSIPS 1.6 RTPProxy + OpenSIPS 1.7 admin 2014-03-07 45311
37 Installing RTPproxy Start RTPproxy in Bridged mode very good admin 2014-03-07 111139
36 OpenSIPS Control Panel (OCP) Installation Guide admin 2014-03-06 290702
35 OpenSIPS Control Panel install guide admin 2014-03-06 103040
34 rtpproxy Module admin 2014-03-06 43788
33 MediaProxy Installation Guide admin 2014-03-06 191413
32 How to install OpenSIPS on CentOS debian module add xcap admin 2014-03-06 50645
31 Problem with presence_xml module Opensips 1.9 admin 2014-03-06 52415
30 Building Telephony Systems with OpenSIPS 1.6 books file admin 2014-03-06 47254
29 Multimedia Service Platform admin 2014-03-06 41957
28 How to install OpenSIPS on CentOS Debian etc admin 2014-03-05 49574
27 Opensips Installation, How to. admin 2014-03-05 80559
26 100% CPU usage opensips admin 2014-03-05 59838
25 A2Billing and OpenSIPS admin 2014-03-04 46764
24 Opensips_1.9 install guide this is great I like this admin 2014-03-04 114268
23 Opensips install debian admin 2014-03-03 42838